Speech encoding and decoding with pitch filter range unrestricted by codebook range and preselecting, then increasing, search candidates from linear overlap codebooks

• US 5,819,213 A
• Filed: 01/30/1997
• Issued: 10/06/1998
• Est. Priority Date: 01/31/1996
• Status: Expired due to Term

• Current Assignee: Kabushiki Kaisha Toshiba (Toshiba Corporation)
• Original Assignee: Kabushiki Kaisha Toshiba (Toshiba Corporation)
• Inventors: Oshikiri, Masahiro, Miseki, Kimio, Akamine, Masami, Amada, Tadashi
• Primary Examiner(s): Hudspeth, David R.
• Assistant Examiner(s): Storm, Donald L.
• Application Number:
  US08/791,741
• Time in Patent Office: 614 Days
• Field of Search: 704/200, 704/201, 704/207, 704/219-223
• US Class Current: 704/222
• CPC Class Codes: G10L 19/08 Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
• CPC Class Codes: G10L 2019/0011 Long term prediction filters, i.e. pitch estimation

Abstract

A speech encoding method and apparatus including analyzing, using a codebook expressing speech parameters within a predetermined search range, an input speech signal in an audibility weighting filter corresponding to a pitch period longer than the search range of the codebook, and searching, from the codebook, on the basis of the analysis result, a combination of speech parameters by which the distortion of the input speech signal is minimized, and encoding the combination. The apparatus uses an adaptive codebook of pitch and a noise codebook. The codebooks search a group formed by extracting vectors of predetermined length from one original code vector, while sequentially shifting position so that the vectors overlap each other. The search group is further restricted and another preselection is made before the final search. Search is based on inversely convoluted, orthogonally transformed vectors.

Claims

1. A speech encoding method using a codebook expressing speech parameters within a predetermined search range, comprising:

analyzing an input speech signal in an audibility weighting filter corresponding to a pitch period longer than the search range of the codebook; and

searching, from the codebook, on the basis of the analysis result, a combination of speech parameters by which distortion is minimized, and encoding the combination.

2. A method according to claim 1, wherein the codebook uses an adaptive codebook expressing a plurality of pitch periods within a predetermined search range and a noise codebook expressing a noise string within a predetermined number of candidates, and the searching of the codebook includes searching the adaptive codebook and the noise codebook on the basis of the analysis result and combining a pitch period and a noise string by which the distortion is minimized.

3. A method according to claim 1, wherein the analyzing of an input speech signal includes using the audibility weighting filter and setting a transfer function of the audibility weighting filter on the basis of an LPC coefficient obtained by performing LPC analysis for an input speech signal and a pitch period and a pitch filter coefficient obtained by analyzing the input speech signal in units of frames, and filtering the input speech signal in accordance with the transfer function.

4. A method according to claim 3, further comprising calculating a prediction residual error signal of the input speech signal by using the LPC coefficient, calculating, on the basis of a signal obtained by multiplying the prediction residual error signal by a Hamming window, an autocorrelation value within a predetermined pitch period analysis range, calculating a pitch period at which the autocorrelation value is a maximum, and calculating the pitch filter coefficient from the prediction residual error signal and the pitch period.

5. A speech encoding method comprising:

analyzing a pitch period of an input speech signal and supplying the pitch period of the input speech signal to a pitch filter which suppresses a pitch period component;

setting an analysis range of the pitch period to be supplied to the pitch filter so that the analysis range is wider than a range of a pitch period which can be expressed by encoded data of a pitch period stored in a codebook; and

searching the pitch period of the input speech signal from the codebook on the basis of a result of analysis performed for the input signal by an audibility weighting filter including the pitch filter, and encoding the pitch period.

6. A method according to claim 5, wherein assuming that the range of the pitch period (TL) which can be expressed by the encoded data is TLL≦

TL≦

TLH and the analysis range of the pitch period (TW) to be supplied to the pitch filter is TWL≦

TW≦

TWH, at least one of conditions TLL>

TWL and TLH<

TWH is met.

7. A speech encoding apparatus comprising:

a codebook expressing speech parameters within a predetermined search range;

an audibility weighting filter for analyzing an input speech signal on the basis of an analysis range of pitch period which is wider than the search range of the codebook; and

an encoder for searching, from the codebook, on the basis of the analysis result, a combination of speech parameters by which distortion is minimized, and encoding the combination.

8. An apparatus according to claim 7, wherein the codebook has an adaptive codebook expressing a plurality of pitch periods within a predetermined search range and a noise codebook expressing a noise string within a predetermined number of candidates, and the encoder comprises means for searching the adaptive codebook and the noise codebook on the basis of the analysis result and combining a pitch period and a noise string by which the distortion is minimized.

9. An apparatus according to claim 7, wherein the audibility weighting filter comprises a filter for setting a transfer function on the basis of an LPC coefficient obtained by performing LPC analysis for an input speech signal and a pitch period and a pitch filter coefficient obtained by analyzing the input speech signal in units of frames, and filtering the input speech signal in accordance with the transfer function.

10. An apparatus according to claim 9, further comprising a calculator for calculating a prediction residual error signal of the input speech signal by using the LPC coefficient, a pitch period analyzer for calculating, on the basis of a signal obtained by multiplying the prediction residual error signal by a Hamming window, an autocorrelation value within a predetermined pitch period analysis range, and calculating a pitch period at which the autocorrelation value is a maximum, and a pitch filter coefficient analyzer for calculating the pitch filter coefficient from the prediction residual error signal and the pitch period.

11. A speech encoding apparatus comprising:

a pitch filter which suppresses a pitch period component of a speech signal;

means for analyzing a pitch period of an input speech signal and supplying the pitch period of the input speech signal to the pitch filter;

means for setting an analysis range of the pitch period to be supplied to the pitch filter so that the analysis range is wider than a range of a pitch period which can be expressed by encoded data of a pitch period stored in a codebook; and

means for searching the pitch period of the input speech signal from the codebook on the basis of a result of analysis performed for the input signal by an audibility weighting filter including the pitch filter, and encoding the pitch period.

12. An apparatus according to claim 11, wherein assuming that the range of the pitch period (TL) which can be expressed by the encoded data is TLL≦

TL≦

TLH and the analysis range of the pitch period (TW) to be supplied to the pitch filter is TWL≦

TW≦

TWH, at least one of conditions TLL>

TWL and TLH<

TWH is met.

13. A speech decoding method comprising:

analyzing a pitch period of a decoded speech signal obtained by decoding encoded data;

passing the decoded speech signal through a post filter including a pitch filter for emphasizing a pitch period component of the decoded speech signal; and

setting an analysis range of the pitch period to be supplied to the pitch filter so that the analysis range is wider than a range of a pitch period which can be expressed by the encoded data.

14. A method according to claim 13, wherein assuming that the range of the pitch period (TL) which can be expressed by the encoded data is TLL≦

TL≦

TLH and the analysis range of the pitch period (TP) to be supplied to the pitch filter is TPL≦

TP≦

TPH, at least one of conditions TLL>

TPL and TLH<

TPH is met.

15. A speech decoding apparatus comprising:

means for analyzing a pitch period of a decoded speech signal obtained by decoding encoded data;

a post filter including a pitch filter for emphasizing a pitch period component of the decoded speech signal; and

means for setting an analysis range of the pitch period to be supplied to the pitch filter so that the analysis range is wider than a range of a pitch period which can be expressed by the encoded data.

16. An apparatus according to claim 15, wherein assuming that the range of the pitch period (TL) which can be expressed by the encoded data is TLL≦

TL≦

TLH and the analysis range of the pitch period (TP) to be supplied to the pitch filter is TPL≦

TP≦

TPH, at least one of conditions TLL>

TPL and TLH<

TPH is met.

17. A vector quantization method comprising:

selecting, as pre-selecting candidates, a plurality of code vectors relatively close to a target vector from a predetermined code vector group;

generating expanded pre-selecting candidates by restricting selection objects for the pre-selecting candidates to some code vectors of the code vector group, selecting some code vectors other than the selection objects from the code vector group on the basis of the pre-selecting candidates and adding the selected code vectors as new pre-selecting candidates; and

searching an optimum code vector closer to the target vector from the expanded pre-selecting code vectors.

18. A vector quantization method comprising:

selecting, as pre-selecting candidates, a plurality of code vectors relatively close to a target vector from a code vector group formed by extracting code vectors of a predetermined length from one original code vector while sequentially shifting positions of the code vectors such that adjacent code vectors overlap each other;

generating expanded pre-selecting candidates by restricting selection objects for the pre-selecting candidates to some code vectors positioned at predetermined intervals in the code vector group and adding code vectors in the code vector group, other than the selection objects and positioned near the pre-selecting candidates, as new pre-selecting candidates; and

searching an optimum code vector closer to the target vector from the expanded pre-selecting candidates.

19. A speech encoding method comprising:

generating a drive signal by using an adaptive code vector and a noise code vector;

supplying the drive signal to a synthesis filter whose filter coefficient is set on the basis of an analysis result of an input speech signal, thereby generating a synthesis speech vector;

searching an optimum adaptive code vector and an optimum noise code vector for generating a synthesis speech vector close to a target vector calculated from the input speech signal from a predetermined adaptive code vector group and a predetermined noise code vector group, respectively;

orthogonally transforming the target vector with respect to the optimum adaptive code vector convoluted by the synthesis filter and inversely convoluting the target vector by the synthesis filter, thereby generating an inversely convoluted, orthogonally transformed target vector;

restricting some noise code vectors in the noise code vector group as selection objects for pre-selecting candidates;

calculating evaluation values relating to distortions of the noise code vectors as the selection objects with respect to the inversely convoluted, orthogonally transformed target vector, and selecting the pre-selecting candidates from the selection object noise code vectors on the basis of the evaluation values;

selecting, on the basis of the pre-selecting candidates, some noise code vectors other than the selection objects from the noise code vector group and adding the selected noise code vectors to the pre-selecting candidates, thereby generating expanded pre-selecting candidates; and

searching the optimum noise code vector from the expanded pre-selecting candidates.

20. A vector quantization method comprising:

weighting each code vector of a code vector group formed by cutting out code vectors of a predetermined length from one original code vector while sequentially shifting positions of the code vectors such that adjacent code vectors overlap each other;

inversely convoluting a target vector of the weighted code vectors and inversely convoluting the original code vector by using the inversely convoluted target vector as a filter coefficient, thereby calculating evaluation values related to distortions with respect to the target vector; and

searching a code vector relatively close to the target vector from the code vector group on the basis of the evaluation values.

21. A vector quantization method comprising:

weighting each code vector of a code vector group formed by extracting code vectors of a predetermined length from one original code vector while sequentially shifting positions of the code vectors such that adjacent code vectors overlap each other;

inversely convoluting a target vector of the weighted code vectors and inversely convoluting the original code vector by using the inversely convoluted target vector as a filter coefficient, thereby calculating evaluation values related to distortions with respect to the target vector; and

selecting, as pre-selecting candidates, a plurality of code vectors relatively close to the target vector from the code vector group on the basis of the evaluation values, and searching an optimum code vector closer to the target vector from the pre-selecting candidates.

22. A speech encoding method comprising:

generating a drive signal by using an adaptive code vector and a noise code vector;

supplying the drive signal to a synthesis filter whose filter coefficient is set on the basis of an analysis result of an input speech signal, thereby generating a synthesis speech vector;

searching an optimum adaptive code vector and an optimum noise code vector for generating a synthesis speech vector close to a target vector calculated from the input speech signal from a predetermined adaptive code vector group and a noise code vector group formed by cutting out code vectors of a predetermined length from one original code vector while sequentially shifting positions of the code vectors such that adjacent noise code vectors overlap each other, respectively;

orthogonally transforming the target vector with respect to the optimum adaptive code vector convoluted by the synthesis filter and inversely convoluting the target vector by the synthesis filter, thereby generating an inversely convoluted, orthogonally transformed target vector;

inversely convoluting the original code vector with the inversely convoluted, orthogonally transformed target vector, calculating evaluation values related to distortions of the noise code vectors with respect to the inversely convoluted, orthogonally transformed target vector from the inversely convoluted original code vector, and selecting pre-selecting candidates from the noise code vector group on the basis of the evaluation values; and

searching the optimum noise code vector from the pre-selecting candidates.